12-17-2008 08:42 AM - edited 03-15-2019 03:07 PM
Hi All,
I'm a problem using conference with calls between an internal 7931 IP Phone and two external GSM Phone.
To call gsm phones I use an IP GSM gateway (Quescom) with four simultaneous channels (4 GSM SIM)
Normally I haven't any problems when 1, 2, 3 or 4 users call external consumers using GSM gateway.
Now, if I try to start a conference: first call to outgoing user on GSM (for example: 03381234567), next press CONF on IPPhone display, next I try to digit
second GSM phone number, but on phone display appears "impossible to complete conference".
Note: for emergency calls on GSM numbers, (GSM gateway out of service or bad signal cell, etc), I have attivated to force outgoing calls on GSM numbers,
a numbers sequence to digit before to digit GSM number = 0 + 9 + GSM phone number. This mode enable user to call a GSM number using ISDN channels.
Using this method the conference between two external numbers and one internal works correctly, as between tree internal numbers too.
In attach the configuration on the router ISR 2821.
Please could you help me to solve this problem?
Thank you for best support
Kind Regards
Tonino
!
voice call send-alert
voice call convert-discpi-to-prog
voice call carrier capacity active
voice rtp send-recv
!
voice service pots
!
voice service voip
allow-connections h323 to h323
no supplementary-service h450.2
no supplementary-service h450.3
h323
!
voice class codec 10
codec preference 10 g711ulaw
codec preference 13 g711alaw
codec preference 14 g729r8
!
voice class h323 1
h225 timeout tcp establish 3
!
telephony-service
load 7931 SCCP31.8-2-2SR1S
max-ephones 58
max-dn 192
ip source-address 10.70.11.11 port 2000
timeouts interdigit 7
system message COMPANY
url directories http://10.70.10.30/fw/apps/speedy/xml/directory.asp
url authentication http://10.70.10.30/fw/authenticate.asp
user-locale IT
network-locale IT
time-zone 23
time-format 24
date-format dd-mm-yy
voicemail 9999
max-conferences 8 gain -6
call-forward pattern .T
moh music-on-hold.au
transfer-system full-consult dss
transfer-pattern .T
transfer-pattern 0T
secondary-dialtone 0
xmltest
create cnf-files version-stamp 7960 Dec 13 2008 17:03:31
!
dial-peer voice 50 voip
corlist outgoing GSM CALL
description ** DIAL PEER CALL TO GSM GATEWAY **
translation-profile outgoing OUT-GSM-GW
preference 1
max-conn 4
destination-pattern 03T
voice-class codec 10
session target ipv4:10.70.11.12
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 70 pots
corlist outgoing CALL-CELLULARI
description *** OUTGOING FORCED CALLS CELLULARI TO ISDN***
preference 1
destination-pattern 093T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
translate-outgoing calling 1
direct-inward-dial
port 0/0/0
prefix 3
!
dial-peer voice 71 pots
corlist outgoing CALL-CELLULARI
description *** OUTGOING FORCED CALLS CELLULARI TO ISDN***
preference 2
destination-pattern 093T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
translate-outgoing calling 1
direct-inward-dial
port 0/0/1
prefix 3
!
dial-peer voice 72 pots
corlist outgoing CALL-CELLULARI
description *** OUTGOING FORCED CALLS CELLULARI TO ISDN***
preference 3
destination-pattern 093T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
translate-outgoing calling 1
direct-inward-dial
port 0/1/0
prefix 3
!
dial-peer voice 73 pots
corlist outgoing CALL-CELLULARI
description *** OUTGOING FORCED CALLS CELLULARI TO ISDN***
preference 4
destination-pattern 093T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
translate-outgoing calling 1
direct-inward-dial
port 0/1/1
prefix 3
!
12-24-2008 07:17 AM
This could indicate a codec mismatch exists between IP phones used for conferencing.
Do one of the following:
Use the same codecs on phones used for conferencing, and make sure that all of the regions and conference bridges use the same codec. Use hardware conference bridges which can perform transcoding.
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